Coding process for inserting an inaudible data signal into an audio signal, decoding process, coder and decoder

ABSTRACT

In a coding method and a coder for introducing a non-audible data signal into an audio signal, the audio signal is first transformed to a spectral range and the masking threshold of the audio signal is determined. A pseudo-noise signal and a data signal are provided and multiplied with each other so a to provide a frequency-spread data signal. The spread data signal is weighted with the masking threshold, and thereafter the audio signal and the weighted data signal are superimposed. In a method and a decoder for decoding a data signal introduced into an audio signal in non-audible manner, the audio signal is first sampled and thereafter the sampled audio signal is filtered in non-recursive manner. The filtered audio signal is subsequently compared to a threshold value so as to retrieve the data signal.

CROSS REFERENCE TO RELATED APPLICATIONS

This a 371 of PCT/EP97/00338 filed on Jan. 24, 1997.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The present invention relates to a coding method and to a coder forintroducing a non-audible data signal into an audio signal.

2. Description of the Related Art

The transmission of non-audible data signals in an audio signal isemployed for example in range research for broadcasting. Range researchserves to reliably determine the listener distribution of individualradio stations. The prior art knows various solutions for ascertainingthe listener distribution of individual radio stations.

A first method operates such that a microphone, carried by a listener,is used for recording ambient noise which is compared by means of areference receiver. On the basis of this comparison it is possible todetermine the receiving frequency of the radio receiver.

A second method records the ambient noise in compressed form along withthe information of the exact time in a memory and then transmits thesame to a central station. In the latter, the data are compared bypowerful computers with program examples recorded during a predeterminedperiod of time, for example a day. The station listened to can beascertained in this manner.

The methods described hereinbefore display the following deficiencies.

The system described first is not applicable to multi-band reception,multi-standard reception or multi-media reception, since it isrestricted to the transmission of frequency-modulated signals only.Additional local broadcasting of other media via free FM channels ispossible in individual cases only due to the multiplicity of programsources. Furthermore, with this method the same receiving strength asthat of the receiver of the listener is necessary. In case of goodreceiving equipment or e. g. in cars, this requirement cannot befulfilled. Another disadvantage consists in the reaction time for tuningthe reference receiver and the correlation, since this increases withthe numbers of programs offered and is in the range of minutes. Thecurrent consumption of such a method is considerable due to thecomponents used, the receiver, signal processing etc. Moreover, thereceiver cannot be designed in any economic manner desired, since thecurrent consumption of the reference receiver directly determines thelarge-signal strength. Again another disadvantage consists in that thecomparison principle is capable only of determining the frequency of thesignal received, with the frequency occupancy, however, being dependentupon the momentary location. It is thus necessary to obtain informationconcerning the location of the listener, for example via the currenttransmitter tables.

The second method described hereinbefore involves the disadvantage of aconsiderable memory need since in case of recording over 24 hours, a netdata quantity of about 150 MB results. Even in case of good compressione.g. by the factor of 10, a data amount of about 15 MB arises each day.The memories to be utilized are thus large and consequently expensive,and they also have a high current demand. In addition thereto, thedetermination of the reference programs causes difficulties since thisneeds to be performed in distributed manner all over the country. Stillanother problem consists in the problematic nature concerning dataprotection, as the audio information is collected directly from theenvironment of the test person and is conveyed further to a centralevaluation.

For avoiding the problems outlined hereinbefore, the prior art hasalready suggested several methods in which an identification signal of astation is introduced in the form of a data signal into the audio signalto be transmitted. The data signal to be transmitted in this case is notaudible for the listener.

Such methods are described for example in WO 94/11989, GB 2260246 A, GB2292506 A and WO 95/04430. The disadvantage of these methods consists inthat it cannot be ensured that the data signal is not audible to thelistener at all times during transmission of the audio signal.

U.S. Pat. No. 5,450,490 describes an apparatus for and a method ofembedding codes in audio signals and decoding the same. This systemmakes use of various symbols that are coded by means of interleavedfrequency lines. To ensure that the data signals transmitted are notaudible at any time, a masking assessment is carried out with respect tothe individual frequencies of which the symbols to be transmitted arecomposed. The disadvantage of this method consists in that thegeneration of signals to be transmitted is very complex.

U.S. Pat. No. 5,473,631 refers to a communication system fortransmitting at the same time data and audio signals via a conventionalaudio communication channel, making use of psycoacoustic codingtechniques (perceptual coding). A first network is used which monitorsthe audio channel for detecting possibilities for introducing the datasignal into the audio channel in such a manner that the signalsintroduced are masked by the audio signal. There is provided a controlby means of which a data signal is provided which thereafter is storedin RAM memories. The data signal is coded either by a spread-spectrumcoder. The data signal stored in the RAM memory is entered into amodulo2-coder in which it is mixed with a synchronous pseudo-noise codefrom a PN code generator. The resulting signal is introduced into a headsignal generator, and the signal output from this generator is appliedto an adjustable attenuation member. The output of the adjustableattenuation member is connected to a summer which serves to combine theaudio signal and the data signal so as to issue the audio and datasignal at the output thereafter. The network is used for establishingpossibilities of introducing a data signal into the audio signal in sucha manner that the data signals are not perceived by a human listener.

SUMMARY OF THE INVENTION

The object of the present invention resides in providing a method ofcoding a data signal contained in an audio signal in non-audible manner,in which it is ensured that the data signal to be transmitted is notperceptible to the human ear, and which is not susceptible with respectto interference phenomena and establishes good channel exploitationwhile permitting safe and simple decoding of the data signal.

According to a first aspect, the present invention is a coding methodfor introducing a non-audible data signal into an audio signal. Themethod has the following steps:

a) transforming the audio signal to the spectral range;

b) determining the spectrum of the masking threshold exclusively on thebasis of the audio signal;

c) providing a pseudo-noise signal;

d) providing the data signal;

e) multiplying the pseudo-noise signal by the data signal so as toprovide a frequency-spread data signal;

f) weighting the spectrum of the spread data signal with the spectrum ofthe masking threshold;

g) transforming the weighted data signal to the time domain; and

h) superimposing the audio signal and the weighted signal.

According to a second aspect, the present invention is a coding methodfor introducing a non-audible data signal into an audio signal, themethod having the following steps:

a) transforming the audio signal to the spectral range;

b) determining the spectrum of the masking threshold exclusively on thebasis of the audio signal;

c) providing a pseudo-noise signal;

d) providing the data signal;

e) multiplying the pseudo-noise signal by the data signal so as toprovide a frequency-spread data signal;

f) weighting the spectrum of the spread data signal with the maskingthreshold;

g) superimposing the audio signal and the weighted signal in thespectral range; and

h) transforming the weighted data signal to the time domain.

Another object of the present invention resides in providing a coder forintroducing and extracting a data signal contained in an audio signal innon-audible manner, in which it is ensured that the data signal to betransmitted is not perceived by the human ear, and which is notsusceptible with respect to interference phenomena and establishes goodchannel exploitation while permitting safe and simple decoding of thedata signal.

The present invention provides a coder for introducing a non-audibledata signal into an audio signal, having

a means for transforming the audio signal to the spectral range;

a means for determining the spectrum of the masking thresholdexclusively on the basis of the audio signal;

a pseudo-noise signal source;

a data signal source;

a means for multiplying the pseudo-noise signal by the data signal so asto provide a frequency-spread data signal;

a means for weighting the spectrum of the spread data signal with thespectrum of the masking threshold;

a means for transforming the weighted signal to the time domain; and

a means for superimposing the audio signal and the weighted data signal.

The present invention further provides a coder for introducing anon-audible data signal into an audio signal, having

a means for transforming the audio signal to the spectral range;

a means for determining the spectrum of the masking thresholdexclusively on the basis of the audio signal;

a pseudo-noise signal source;

a data signal source;

a means for multiplying the pseudo-noise signal by the data signal so asto provide a frequency-spread data signal;

a means for weighting the spectrum of the spread data signal with themasking threshold;

a means for superimposing the audio signal and the weighted data signalin the spectral range; and

a means for transforming the weighted signal to the time domain.

An advantage of the method according to the invention consists in thatinformation is introduced into an audio signal without being perceivedby the human ear, while however being safely decoded by a detector. Afurther advantage of the present invention resides in thatspread-spectrum-modulation is employed in which the information or datasignal is spread to the entire transmission band, thereby reducing thesusceptibility to interference phenomena and multipath propagation. Atthe same time, good channel exploitation is achieved.

In accordance with the present invention, non-audibility is obtained inthat the audio signal, being for example a music signal, to which thedata signal or information is to be added, is subjected topsychoacoustics calculation. On the basis thereof, the masking thresholdis ascertained, and the spread-spectrum signal is weighted therewith.This ensures that there is at no time more energy used for datatransmission than is admissible psychoacoustically.

The method of decoding the coded data signal makes use of anon-recursive filter (matched filter). The advantage hereof is that thisfilter can be employed for correlation and reconstruction so that themethod of decoding is particularly simple, which is advantageous withrespect to a subsequent hardware realization. A decoder can be provided,for example, in the form of a wrist watch that is easy to wear for testpersons.

An advantage of the coder according to the invention is that informationis introduced into an audio signal without being perceived by the humanear, while however being safely decoded by a detector. A furtheradvantage of the present invention consists in that spread-spectrummodulation is employed in which the information or data signal is spreadto the entire transmission band thereby reducing the susceptibility tointerference phenomena and multipath propagation. At the same time, goodchannel exploitation is achieved.

In accordance with the present invention, the non-audibility is obtainedin that the audio signal, being for example a music signal, to which thedata signal or information is to be added, is subjected topsychoacoustics calculation. On the basis thereof, the masking thresholdis ascertained, and the spread-spectrum signal is weighted therewith.This ensures that there is at no time more energy used for datatransmission than is admissible psychoacoustically.

The decoder makes use of a non-recursive filter (matched filter). Theadvantage hereof resides in that this filter can be employed forcorrelation and reconstruction so that the method of decoding isparticularly simple, which is advantageous with respect to a subsequenthardware realization.

According to a another aspect, the present invention provides anapparatus for determining the listener distribution of individual radiostations by way of an identification signal, the apparatus having acoder which introduces the identification signal into the audio signaland has the following features:

a means for transforming the audio signal to the spectral range;

a means for determining the spectrum of the masking thresholdexclusively on the basis of the audio signal;

a pseudo-noise signal source;

a data signal source;

a means for multiplying the pseudo-noise signal by the data signal so asto provide a frequency-spread data signal;

a means for weighting the spectrum of the spread data signal with thespectrum of the masking threshold;

a means for transforming the weighted data signal to the time domain;and

a means for superimposing the audio signal and the weighted data signal;

and comprising a decoder which extracts the identification signal fromthe audio signal transmitted.

According to a another aspect, the present invention provides anapparatus for determining the listener distribution of individual radiostations by way of an identification signal, the apparatus having acoder which introduces the identification signal into the audio signaland has the following features:

a means for transforming the audio signal to the spectral range;

a means for determining the spectrum of the masking thresholdexclusively on the basis of the audio signal;

a pseudo-noise signal source;

a data signal source;

a means for multiplying the pseudo-noise signal by the data signal so asto provide a frequency-spread data signal;

a means for weighting the spectrum of the spread data signal with themasking threshold;

a means for superimposing the audio signal and the weighted data signalin the spectral range; and

a means for transforming the superimposed signal to the time domain;

and comprising a decoder which extracts the identification signal fromthe audio signal transmitted.

According to a another aspect, the present invention provides anapparatus for determining the transmitter reach of a radio station byway of an identification signal, the apparatus having a coder whichintroduces the identification signal into the audio signal and has thefollowing features:

a means for transforming the audio signal to the spectral range;

a means for determining the spectrum of the masking thresholdexclusively on the basis of the audio signal;

a pseudo-noise signal source;

a data signal source;

a means for multiplying the pseudo-noise signal by the data signal so asto provide a frequency-spread data signal;

a means for weighting the spectrum of the spread data signal with thespectrum of the masking threshold;

a means for transforming the weighted signal to the time domain; and

a means for superimposing the audio signal and the weighted data signalin the spectral range;

and comprising a decoder which extracts the identification signal fromthe audio signal transmitted.

According to a another aspect, the present invention provides anapparatus for determining the transmitter reach of a radio station byway of an identification signal, the apparatus having a coder whichintroduces the identification signal into the audio signal and has thefollowing features:

a means for transforming the audio signal to the spectral range;

a means for determining the spectrum of the masking thresholdexclusively on the basis of the audio signal;

a pseudo-noise signal source;

a data signal source;

a means for multiplying the pseudo-noise signal by the data signal so asto provide a frequency-spread data signal;

a means for weighting the spectrum of the spread data signal with themasking threshold;

a means for superimposing the audio signal and the weighted data signalin the spectral range; and

a means for transforming the weighted signal to the time domain;

and comprising a decoder which extracts the identification signal fromthe audio signal transmitted.

According to a another aspect, the present invention provides anapparatus for identifying audio signals with an unequivocalidentification number for identifying the sources of copies of soundcarriers, the apparatus having a coder which introduces theidentification signal into the audio signal and has the followingfeatures:

a means for transforming the audio signal to the spectral range;

a means for determining the spectrum of the masking thresholdexclusively on the basis of the audio signal;

a pseudo-noise signal source;

a data signal source;

a means for multiplying the pseudo-noise signal by the data signal so asto provide a frequency-spread data signal;

a means for weighting the spectrum of the spread data signal with thespectrum of the masking threshold;

a means for transforming the weighted signal to the time domain; and

a means for superimposing the audio signal and the weighted data signal;

and comprising a decoder which extracts the identification signal fromthe audio signal transmitted.

According to a another aspect, the present invention provides anapparatus for identifying audio signals with an unequivocalidentification number for identifying the sources of copies of soundcarriers, the apparatus having a coder which introduces theidentification signal into the audio signal and has the followingfeatures:

a means for transforming the audio signal to the spectral range;

a means for determining the spectrum of the masking thresholdexclusively on the basis of the audio signal;

a pseudo-noise signal source;

a data signal source;

a means for multiplying the pseudo-noise signal by the data signal so asto provide a frequency-spread data signal;

a means for weighting the spectrum of the spread data signal with themasking threshold;

a means for superimposing the audio signal and the weighted data signalin the spectral range; and

a means for transforming the weighted signal to the time domain;

and comprising a decoder which extracts the identification signal fromthe audio signal transmitted.

According to a another aspect, the present invention provides anapparatus for the remote control of audio apparatus by way of a controlsignal, the apparatus having a coder which introduces the control signalinto the audio signal and has the following features:

a means for transforming the audio signal to the spectral range;

a means for determining the spectrum of the masking thresholdexclusively on the basis of the audio signal;

a pseudo-noise signal source;

a data signal source;

a means for multiplying the pseudo-noise signal by the data signal so asto provide a frequency-spread data signal;

a means for weighting the spectrum of the spread data signal with thespectrum of the masking threshold;

a means for transforming the weighted signal to the time domain; and

a means for superimposing the audio signal and the weighted data signal;

and comprising a decoder which extracts the identification signal fromthe audio signal transmitted.

According to a another aspect, the present invention provides anapparatus for the remote control of audio apparatus by means of acontrol signal, the apparatus having a coder which introduces thecontrol signal into the audio signal and has the following features:

a means for transforming the audio signal to the spectral range;

a means for determining the spectrum of the masking thresholdexclusively on the basis of the audio signal;

a pseudo-noise signal source;

a data signal source;

a means for multiplying the pseudo-noise signal by the data signal so asto provide a frequency-spread data signal;

a means for weighting the spectrum of the spread data signal with themasking threshold;

a means for superimposing the audio signal and the weighted data signalin the spectral range; and

a means for transforming the weighted signal to the time domain;

and comprising a decoder which extracts the identification signal fromthe audio signal transmitted.

According to a another aspect, the present invention provides anapparatus for providing a data channel of low bit rate in digitallyoperating audio apparatus, the data channel operating in parallel to theaudio signal, the apparatus having a coder which introduces theinformation into the audio signal and has the following features:

a means for transforming the audio signal to the spectral range;

a means for determining the spectrum of the masking thresholdexclusively on the basis of the audio signal;

a pseudo-noise signal source;

a data signal source;

a means for multiplying the pseudo-noise signal by the data signal so asto provide a frequency-spread data signal;

a means for weighting the spectrum of the spread data signal with thespectrum of the masking threshold;

a means for transforming the weighted signal to the time domain; and

a means for superimposing the audio signal and the weighted data signal;

and comprising a decoder which extracts the identification signal fromthe audio signal transmitted.

According to a another aspect, the present invention provides anapparatus for providing a data channel of low bit rate in digitallyprocessing audio apparatus, the data channel operating in parallel tothe audio signal, the apparatus having a coder which introduces theinformation into the audio signal and has the following features:

a means for transforming the audio signal to the spectral range;

a means for determining the spectrum of the masking thresholdexclusively on the basis of the audio signal;

a pseudo-noise signal source;

a data signal source;

a means for multiplying the pseudo-noise signal by the data signal so asto provide a frequency-spread data signal;

a means for weighting the spectrum of the spread data signal with themasking threshold;

a means for superimposing the audio signal and the weighted data signalin the spectral range; and

a means for transforming the weighted signal to the time domain;

and comprising a decoder which extracts the identification signal fromthe audio signal transmitted.

BRIEF DESCRIPTION OF THE DRAWINGS

In the following, preferred embodiments of the present invention will beelucidated in more detail by way of the accompanying drawings in which

FIG. 1 shows an embodiment of a coder according to the invention;

FIG. 2 is a representation of a transmission frame used for transmittingthe useful signal;

FIG. 3 is a block diagram of the source coding block shown in FIG. 1;

FIG. 4 shows an embodiment of a decoder according to the invention;

FIG. 5 is a block diagram of the data decoder shown in FIG. 4;

FIG. 6 shows an embodiment of a system for determining the listenerdistribution of a radio station, making use of the coding and decodingmethods according to the invention;

FIG. 7 shows an embodiment of a system for determining the listenerdistribution of a radio station, making use of the coding and decodingmethods according to the invention;

FIG. 8 shows an embodiment of a system for identifying audio signalswith an unequivocal identification number for identifying soundcarriers; and

FIG. 9 shows an embodiment of a system for remote control of audioequipment, making use of the coding and decoding methods according tothe invention.

DESCRIPTION OF THE PREFERRED EMBODIMENTS

In the following, an embodiment of a coder will be described in moredetail with reference to FIG. 1. It is to be understood that the circuitshown in FIG. 1 constitutes merely a preferred embodiment, without thepresent invention being restricted thereto.

The coding circuit depicted in FIG. 1 consists of a transformation block100, a psychoacoustics block 102, a data signal generator 104, a sourcecoding block 105, a pseudo-noise signal generator 106, a BPSK basebandmodulator 108 (BPSK=Binary Phase Shift Keying), a BPSK modulator 110, ameans for weighting two signals 112, a retransformation block 114, and asuperposition means 116. In the embodiment shown in FIG. 1, the BPSKbaseband modulator 108, the BPSK modulator 110 and the means forweighting two signals 112 are each constituted by a multiplier.Moreover, an additional transformation block 118 is provided,transforming the output signal s(l) of BPSK modulator 110 to thespectral range.

Transformation block 100 is connected to an input IN of the circuit. Theoutput of transformation block 100 is connected to psychoacoustics block102. The input of the circuit is connected furthermore to an input ofsuperposition means 116.

The output of pseudo-noise signal generator 106 is connected to an inputof BPSK baseband modulator 108, and the output of data signal generator104 is connected to the input of source coding block 105 whose output inturn is connected to the other input of BPSK baseband modulator 108. Theoutput of BPSK baseband modulator 108 is connected to an input of BPSKmodulator 110 having its other input connected to a signal generator(not shown) applying a cosinusoidal signal to the other input of BPSKmodulator 110. The output of BPSK modulator 110 is connected to theadditional transformation block 118 having its output connected toweighting means 112.

The output of psychoacoustics block 102 is also connected to weightingmeans 112. The output of weighting means 112 is connected to an input ofretransformation block 114. The output of retransformation block 114 isconnected to a further input of superposition means 116, with the outputof superposition means 116 being connected to an output OUT of thecircuit.

In the following, a preferred embodiment of the coding method accordingto the invention will be described in more detail by way of FIG. 1.

At first, a music signal n(k) is fed at input “IN”, which is present forexample as digital PCM music signal (PCM=Pulse Coded Modulation). Intransformation block 100, the music signal is first subjected to windowtransformation using a Hamming window and thereafter is transformed tothe spectral range by fast Fourier transform (FFT=Fast Fouriertransform) having a length of 1024 with 50% overlap. Thereafter, thespectrum N(ω) of music signal n(k) is present with 512 frequency lines,which is used as input signal for psychoacoustics 102. The spectrum ofthe music signal is applied at the same time to superposition means 116,as indicated by arrow 120.

In psychoacoustics block 102, the spectrum N(ω) is divided into criticalbands. These bands have a width of ⅓ bark, which depending on thesampling frequency (in the present embodiment, this frequency is e.g.44.1 kHz or 48 kHz) results in a band number of approx. 60 criticalbands. The allocation of the frequencies f(Hz) to bands z(bark) isoriented along the lines of the band partitioning made by the human earduring hearing and is noted, for example, in standard ISO/IEC 11172-3 intable form. In these critical bands, the band energy is determined bysummation of the real part and the imaginary part of the spectrum N(ω)according to the following equation:

E _(i) =Re(N(ω_(i)))² +Im(N(ω_(i)))²

This energy distribution is then subjected to spreading. To this end,the so-called spread function is calculated, using the standard ISO/IEC11172-3 (1993). Thereafter, the 60 spread courses or waveforms obtainedare subjected to convolution with the band energies, thereby obtainingthe excitation course or waveform. On the basis of the latter, it ispossible to calculate the masking threshold W(z) for non-tonal audiosignals in consideration of the masking extent, using one interpolationpoint for each critical band Z.

For tonal audio signals, the masking threshold W(z) is to be ratedconsiderably lower. Thus, with the aid of signal prediction, a measurefor the tonality is determined for each frequency line. The predictiondetermines from the two preceding FFTs for each line a predicted vectorby addition of the difference in phase and amount from the vector of thelast FFT line. Thereafter, an error vector is formed by establishing thedifference between predicted vector and actual vector obtained from theFFT.

By establishing the amount of the error vector in the form of lines, ameasure for the non-predictability of the signal (abbreviated cw=chaosmeasure) for each ω. From this “cw” value, which may take values between0—“very tonal”—and 1—“non-tonal”—, the masking measure can be calculatedthat is to be taken into consideration in calculating the maskingthreshold.

As an alternative, the calculation of the masking threshold can alsotake place in different manner. The spectral lines obtained from FFT arecombined in critical bands. These bands have a width of ⅓ bark, whichdepending on the sampling frequency (in the present embodiment, thisfrequency is e.g. 44.1 kHz or 48 kHz) results in a band number ofapprox. 60 critical bands. The allocation of the frequencies f(Hz) tobands z(bark) is oriented along the lines of the band partitioning madeby the human ear during hearing and is noted, for example, in standardISO/IEC 11172-3 in table form. In these critical bands, the band energyis determined by summation of the real part and the imaginary part ofthe spectrum N(ω) according to the following equation:

E _(i) =Re(N(ω_(i)))² +Im(N(ω_(i)))²

It shall be assumed now that the entire band contains tonal signalsonly. In this case (worst case), the masking threshold results a fixedamount below the energy distribution of the music signal. As maximummasking extent e.g. −18 dB can be assumed. The advantage of this methodconsists in that the calculation is very simple, since neitherconvolutions nor predictions have to be carried out. The disadvantageresides in that energy reserves delivered by the music signal withrespect to masking possibly are not utilized. However, when sufficientprocessing gain has been made available, this disadvantage is notdisturbing.

W(z) then is converted to W(ω), this conversion making use of standardISO/IEC 11172-3. Thus, the waveform of masking threshold W(ω) is appliedto the output of block 102 and in dicates up to which energy level onthe signal energy may be applied at a location ω such that thisalteration remains non-audible.

Data signal generator 104 (DSG) makes available the useful data signalx(n) which as a rule is repeated cyclically for enabling decoding in adecoder at any time. The data signal has a bandwidth of 50 Hz forexample. The data at the output of DSG 104 are in the form of a binarysignal and have a low bit rate 1/Tx in the range of 1-100 bits/s. Thespectrum of this signal must be of very narrow-band type in comparisonwith the spectrum of the signal issued by PN signal generator 106 withω_(x).

The useful data signals x(n) in the embodiment shown in FIG. 1 consistof words having a length of 11 bits. These data words are included in aframe having a length of between 26 and 29 bits. FIG. 2 shows thestructure of such a transmission frame in more detail. Transmissionframe 200 includes four sections 202, 204, 206, 208. The first sectionis a synchronous word 202 consisting of seven bits (bits 0 to 6) andconstituted by the bit sequence 1111110 in the embodiment shown in FIG.2. The second section 202 serves for error protection and consists offour bits (bits 7 to 10). The third section 206 contains the data wordhaving a length of 11 bits (bits 11 to 21). The fourth section 208contains a check sum of four bits (bits 22 to 25).

The error protection (section 204 in FIG. 2) is realized by anon-systematic (15,11)-Hamming code. This block code is suitable forcorrecting all 1-bit errors. In case of multibit errors, the data wordobtained is considered wrong and rejected. The advantage of this code isthat it can be realized without great computer expenditure, by simplematrix multiplication, and thus is suitable also as regards the decodingmethod.

Due to the fact that the transmission channel operates in bit-orientedmanner, the transmission frame has to be transmitted along with a HDLCprotocol (HDLC=high-level data link control). This protocol is modifiedsuch that a “0” is not only inserted after six successive “1” bits, butalso a “1” is inserted after six “0”-bits. This modification isnecessary for recognizing and correcting phase deviations that may occuron the channel.

The transmission frame 200 is established by source coding block 105(FIG. 1). FIG. 3 shows source coding block 105 in detail.

The data signals are made available to source coding block 105 from datasignal generator 104. At the input 302 of block 105, the data arepresent in the form of data words having a length of 11 bits, as shownin FIG. 3. The transmission frame is composed such that error protectionis realized first in a first block 304 by the (15,11)-Hamming code. Theframe now has a length of 15 bits. Thereafter, the check sum is added tothe frame in a second block 306. The length then is 19 bits. In block318, the necessary coding of the transmission frame by a HDLC codertakes place, resulting in a frame length of 19 to 22 bits. The binarysignal present at the output of block 308 then is transformed to anantipodal signal. This can be done e.g. with a relationship 0->1 and1->−1. For completing the frame, the synchronous word is added theretoin block 310. At output 312 of source coding block 105, the transmissionframe is present with a length of 26 to 29 bits, which is fed to BPSKbaseband modulator 108.

Pseudo-noise signal generator 106 (PNSG) provides the spread signal g(l)having the bit rate 1/Tg. The bandwidth ω_(g) of this signal determinesthe bandwidth ω_(s) of the spread-spectrum signal and is in the range of6 kHz in the embodiment shown in FIG. 1. The higher frequencies offeredby a high-grade music signal were disregarded in consideration of thefrequency response of the reproduction equipment (e.g. portable radioreceivers). PNSG 106 according to an embodiment is composed as a fedbackshift register and delivers a pseudo-random pseudo-noise sequence (PNsequence) having a length N. This sequence must be known in the decoderfor decoding the signal.

The ratio Tx/Tn is referred to as spread factor and directly determinesthe signal to noise ratio up to which the method still operates inreliable manner. According to the embodiment described herein, thespread factor is 128 and the signal to noise ratio thus is SIN=10 log10(Tx/Tn)=−21 dB.

The binary signal g(l) provided by PNSG 106 then is converted to anantipodal signal. This may take place e.g. with the relationship 0->1and 1->−1. After such formatting, the signal has been processed and isfed to BPSK baseband modulator.

BPSK baseband modulator 108 is designed in simple manner when antipodalsignals are used, since multiplication by sampling values corresponds toBPSK modulation. The resulting signal h(l)=g(l)x′(n) has a bandwidth ofω_(h)≈6 kHz. The amplitude values are −1 and 1. The signal has its mainmaximum at 0 Hz and thus is present in the baseband.

The baseband signal h(l) now is supplied to BPSK modulator 110. In thelatter, the baseband signal h(l) is modulated onto a cosinusoidalcarrier cos(ω_(T)t). The frequency of the carrier is half of thebandwidth of the spread band signal in the baseband. Thus, the firstzero digit of the modulated spectrum comes to lie at 0 Hz. The signalcan thus be transmitted on channels whose transmission function providesstrong attenuation in the range from 0 to 100 Hz, as expected in audiotransmissions via loudspeaker and microphone.

As an alternative, modulation can take place by suitable coding insteadof a cosinusoidal carrier. Due to the specific property of beingaverage-free, it is also possible to employ the Manchester code. Due tothe average-free design thereof, no energy of the spread-band signal isapplied at 0 Hz either, which is important for transmittability. Thecoding regulation for the Manchester code is 0->10 and 1->01. The numberof the bits is thus doubled.

The time signal s(l) available at the output of BPSK modulator 110 thenis transformed to the spectral range in transformation block 118 bymeans of a fast Fourier transform, so that S(ω) is present at the outputof block 118.

The spectral course or waveform of the spread useful signal S(ω) now isweighted with the course or waveform of masking threshold W(ω) throughweighting block 112, with the result that at no location in the audiospectrum is there more noise energy introduced by the spread-spectrumsignal than is perceptible to the human ear. With respect to thedemodulation of the useful signal, the statically changing course of theenergy distribution in the useful signal is of little effect only, sincethe method is particularly powerful especially in this context.

Thereafter, retransformation takes place through inverse fast Fouriertransform in block 114, so that the coded music signal is again presentin the time domain. The 50% overlap is to be noted in theretransformation.

At block 116, the psychoacoustically weighted useful signal in the timedomain is added to the music signal n(k).

The coder, at the output “OUT”, delivers a digital PCM signal n_(c)(k)that can be transmitted on an arbitrary transmission route as long asthe same has a bandwidth of at least 6 kHz.

As an alternative to the embodiment described hereinbefore, the outputof transformation block 100, instead of the input of the circuit, can beconnected in addition to superposition means 116. In this case, thespectral spread signal and the spectral audio signal are superimposed,whereafter retransformation to the time domain takes place.

In the following, a preferred embodiment of a decoding circuit will bedescribed which is used for performing a preferred embodiment of themethod of decoding a data signal contained in an audio signal innon-audible manner according to the invention.

The decoder comprises a microphone 400 receiving, for example, a musicsignal transmitted from a radio receiver. The output of microphone 400is connected to the input of a lowpass 402 having its output connectedto an amplifier 404 with automatic gain control. The output of amplifier404 is connected to an analog/digital converter 406. The output ofanalog/digital converter 406 is connected to the input of anon-recursive filter 408 (matched FIR-filter) having its outputconnected to an input of a bit synchronization control block 410. Theoutput of block 410 is connected to the input of a data decoder 412. Thedecoded data signal is available at the output of data decoder 412.

In the following, an embodiment of the decoder according to theinvention will be described by way of FIG. 4. The music signal n_(c)(k)broadcast by the radio receiver is converted by microphone 400 intoelectrical signals and fed to lowpass 402. The limit frequency oflowpass 402 is such that the frequency portions having no data modulatedtherein are strongly attenuated. In the present embodiment the limitfrequency is 6 kHz. Lowpass filtering has the function of avoidingoverlap distortions which may occur by the subsequent sampling of thesignal.

Amplifier 404 with automatic gain control (AGC=Automatic Gain Control)ensures a constant instantaneous power of the input signal upstream ofA/D converter 406. This is necessary for being able to compensate fortemporary attenuations due to a particular channel. It is pointed outthat the decoder can be realized both in terms of hardware and in termsof software. In case of a software realization, amplifier 404 can bedispensed with.

The A/D converter carries out sampling and digitization of the signal.

Matched filter 408 consists of a FIR-filter or non-recursive filter.Filter 408 contains as coefficient the inverse sequence of the PNsequence of the transmitter. The PN sequence of the pseudo-noise signalcan be Manchester-coded, for example. In that case, filter 408 containsas coefficient the inverse Manchester-coded sequence of the PN sequenceof the transmitter. With maximum correlation, filter 408 thus produces apeak at the output with a sign corresponding to that of the transmittedsymbol. The filter output, at a distance of the length 2*N of the PNsequence, thus delivers peaks representing the data transmitted. Due tothe fact that the peaks cannot be determined unequivocally at all times,filter 408 has the bit synchronization control block 410 connecteddownstream thereof.

The synchronization control in block 410 searches the output signal offilter 408 for peaks which unequivocally stand out from the noisebackground. Once such a peak has been found, keying is performed intothe output of filter 408 synchronously with the length of the PNsequence, in order to retrieve the symbols transmitted. If anunambiguous peak appears during this time, the sampling time iscorrected in corresponding manner.

The output of block 410 delivers a bit stream that is processed in thesubsequent data decoder 412. This bit stream, in the event that novalidly coded signal is present at the input of microphone 402,constitutes a random sequence of bits. When the decoder isbit-synchronized, the bit stream contains the data transmitted.

In data decoder 412, decoding of the useful signal from the bit streamfrom block 410 takes place. The data decoder will now be described inmore detail with reference to FIG. 5. Data decoder 412 comprises aninput IN connected to a frame synchronization block 502 and a HDLCdecoder block 504. Block 502 outputs a trigger signal to block 504. Theoutput of block 504 is connected to the input of a Hamming errorcorrection block 506 having its output connected to the input of a checksum block 508. Subsequent to block 508, Hamming data calculation takesplace in block 410. The output of block 410 is connected to the outputOUT of data decoder 412 having the data word with a length of 11 bitspresent at its output.

Frame synchronization block 502 receives the input bit stream andsearches therein the synchronization word 202. When the latter is found,HDLC decoder 504 is triggered and the input data are decoded incorresponding manner. Thereafter, syndrome calculation and errorcorrection take place using the Hamming code. By way of thebit-error-corrected 15-bit word, the check sum is calculated andcompared to the bits transmitted. When all of these operations aresuccessful, the 15 bits are decoded using the Hamming code, and the 11data bits transmitted are output from the decoder.

It is pointed out that the coding and decoding methods describedhereinbefore constitute merely preferred embodiments of the presentinvention without intention to restrict the invention thereto.

The essential features of the coding method according to the inventionfor introducing a non-audible data signal into an audio signal aretransforming the audio signal to the spectral range, determining themasking threshold of the audio signal, providing a pseudo-noise signal,providing the data signal, multiplying the pseudo-noise signal by thedata signal so as to provide a frequency-spread data signal, weightingof the spread data signal with the masking threshold, and superimposingthe audio signal and the weighted signal.

The essential features of the method of decoding a data signal containedin an audio signal in non-audible manner, according to the invention,are sampling the audio signal, non-recursive filtering of the sampledaudio signal and comparing the filtered audio signal to a thresholdvalue so as to retrieve the data signal.

In the following, a system according to the present invention fordetermining the listener distribution of individual radio stations byway of an identification signal will be described with reference to FIG.6. The system described by way of FIG. 6 uses the afore-described codingmethod for introducing the identification signal to the audio signaltransmitted and uses the above-described decoding method for decodingthe signal from the audio signal received.

The system described by way of FIG. 6 renders possible to ascertain thelistener distribution of the individual radio stations in reliablemanner. The system is independent of the receiving apparatus employed,so that the different listening habits can be taken into account.

The broadcasting transmission also can take place via different media:

FM (analog)

cable (analog and digital)

DAB (220 MHz terrestrial; 1.5 GHz terrestrial and satellite-based)

ADR

Analog satellites subcarriers (television satellites)

LW/MW/SW

television sound.

It is specific to each country which media are relevant for evaluation,but the system shown in FIG. 6 is capable of supporting the media listedabove. The detection of the listener reach takes place in predeterminedtime intervals which are adjustable depending on each particular case.According to an example, the time interval may be 10 seconds.Furthermore, a definition has to be made as to how current theevaluation has to be. According to the example of a system shown in FIG.6, the listener data are detected during the night. In otherembodiments, it may be sufficient to send in the detection apparatus inintervals of 4 weeks each for data evaluation.

The system as shown in FIG. 6 in more detail comprises a detectionapparatus reaching a high degree of acceptance on the side of thelisteners, so as to ensure the reliability of the data collection. Forproviding an as comprehensive as possible data acquisition, thedetection apparatus is carried on the body of a test listener or testperson, and this detection apparatus is a small apparatus withsufficient battery supply, for example by storage cells, which has apleasing design and is easy to handle. The storage cells are reloaded ina charging or docking station.

The system according to the invention in FIG. 6 in its entirety bearsreference numeral 600. System 600 consists of the following components.An audio signal is generated in a radio station 602 and by means of anidentification generator 604 has an identification signal appliedthereto. The application of the audio signal by identification generator604 takes place using the afore-described coding method for introducinga non-audible data signal into an audio signal. The audio signal havingthe identification signal applied thereto is passed further to anantenna 606 effecting broadcasting 608 of the audio signal. A broadcastreceiver 610 consisting of an antenna 612, a receiver apparatus 614 andtwo loudspeakers 616 receives the broadcast audio signal. The audiosignal received by antenna 612 is converted via receiver 614 andloudspeakers 616 into an audible audio signal 618 which is received by adetection apparatus. In the embodiment shown in FIG. 6, receivingapparatus 620 is in the form of a wrist watch. Detection apparatus 620is effective for extracting the identification signal from the audiosignal 618 received. This takes place with the aid of the methodaccording to the invention for decoding a data signal contained in anaudio in non-audible manner. The identification signal ascertained byreceiving apparatus 620 is latched in the receiving apparatus. There isprovided a so-called docking station for accommodating wrist watch 620for example during the night, so as to effect transmission of theidentification data stored. Docking station 622 can be connected to acommunication network 630, which in an embodiment is the telephonenetwork, via a line 624 and a corresponding connecting means 626 whichmay have a telephone 628 connected thereto in addition. Via thecommunication network 630, the data stored in receiving apparatus 620,i.e. the identification data, are sent to a central station 623 whichcomprises a computer 634 for evaluating the data received. Computer 634is connected via a line 636 to a modem 638 which in turn is connected tocommunication network 630 via a line 640 and an additional connectingmeans 642.

The system depicted in FIG. 6 is capable of reliably ascertaining thelistener data of selected radio stations for the current day, with theresolution of the system in terms of time being in the range of a fewseconds. Due to the technology with little complexity, the same can berealized in inexpensive manner.

In the following, a system according to the present invention fordetermining the transmitter reach of a radio station by way of anidentification signal will be described in more detail with reference toFIG. 7. The system described by way of FIG. 7 uses the afore-describedcoding method for introducing the identification signal to the audiosignal transmitted and uses the above-described decoding method fordecoding the signal from the audio signal received.

The system according to the invention in FIG. 7 in its entirety bearsreference numeral 700. In system 700, an audio signal is generated in aradio station 702, for example in a studio 704, and by means of anidentification generator or coder 706 has an identification signalapplied thereto. The application of the audio signal by identificationgenerator 706 takes place using the afore-described coding method forintroducing a non-audible data signal into an audio signal. The audiosignal having the identification signal applied thereto is passedfurther to an antenna 708 effecting broadcasting 710 of the audiosignal. A broadcast receiver 712, for example a test receiver,consisting of an antenna 714 and a receiver apparatus 716 receives thebroadcast audio signal. The receiver 716 shown in FIG. 7 serves only forreceiving the audio signal. As this embodiment is concerned only withthe determination of the transmitter reach, a reproduction of the audiosignal transmitted can be dispensed with.

An advantage of this procedural mode consists in that, for determiningthe transmitter reach, not only a limited band range in the audio signalcan be used for transmitting the audio signal. Rather, it is possible toutilize the entire bandwidth of the audio signal transmitted. Thispermits an increase either of the decoding safety or of the amount ofdata transmitted.

In the embodiment shown in FIG. 7, decoder 718 performing the decodingmethod is constituted by a computer 720 realizing the method by way ofsoftware technology. As can be seen in FIG. 7, receiver 716 iseffectively connected via a line or cable 722 to a so-called sound card724 in the computer for rendering possible processing of the audiosignal by the computer. The transmission from receiver 712 to decoder718 via line 722 takes place in analog manner. In other words, the audiosignal received is fed directly from receiver 712 to decoder 718.

Decoder 718 is connected via a line 724 to a modem 728 which in turn isconnected to a corresponding connecting means 732 via an additional line730. Connecting means 732 is connected to a communication network 734,for example a telephone network. Via communication network 734, the dataascertained from the data signal, i.e. the identification data, are sentto a central station 736 comprising a computer 738 for evaluating thedata received. Computer 738 is connected via a line 740 to a modem 742which in turn is connected to communication network 734.

In the following, a system for identifying audio signals will bedescribed with reference to FIG. 8, which serves to identify soundcarriers and copies of sound carriers by way of the identificationsignal introduced into the audio signal. The advantage resides in thatit is rendered possible thereby to easily identify possible piratedcopies, since each individual sound carrier is provided with anindividual identification in the factory.

FIG. 8a depicts the production of a sound carrier, such as for example acompact disk “CD”, in a press assembly 800. Press assembly 800 comprisesa reproducing means 802 running a master tape containing the audiosignals to be applied to a CD. The CD is pressed in a press mechanism804. Between press mechanism 804 and reproducing means 802, there isdisposed a coder 806. By means of the coder, each CD has anidentification signal associated therewith which is introduced into theaudio signal. Coding takes place in accordance with the above-describedcoding method. For ensuring the generation of individual identificationsignals for individual CDs, coder 806 has a counter associated therewithwhich, for example, makes available consecutive identification numbersas identification signal for introduction into the audio signal.

On the basis of FIG. 8b, the effect of the identifications on individualCDs shall be elucidated in more detail. A CD 808 provided with anindividual identification is copied several times, as indicated by theschematically shown reproducing apparatus 810. The copies can be madeboth in analog and in digital manner.

After the identification has been introduced into the audio signal, thisidentification is maintained also in case of transmission of the audiosignal in the form of a soundfile via the internet, as indicated bynumeral 812 in FIG. 8. This permits conclusions to be made to thesoundfile on the sound carrier.

In the following, a further embodiment will be described with referenceto FIG. 9. FIG. 9 shows a system for remote control of audio apparatus,which makes use of the coding and decoding methods according to theinvention.

The system according to the invention in FIG. 9 in its entirety bearsreference numeral 900. In this system 900 an audio signal is generatedin a radio station 902, for example in a studio 904. By means of a coder706, a data signal or control signal is introduced into the audiosignal. The application of the audio signal by way of coder 906 takesplace using the afore-described coding method for introducing anon-audible data signal into an audio signal. The audio signal havingthe signal applied thereto is passed on to an antenna 908 effectingbroadcasting 910 of the audio signal. A receiver 912, consisting of anantenna 914 and a receiver apparatus 916, receives the emitted audiosignal. Receiver 916 has a decoder provided therein which extracts thedata signal contained in the audio signal in accordance with thedecoding method described hereinbefore. The receiver is constructed suchthat it is responsive to the data signal, for example, for beginningrecording of a music program of a radio station. Due to the data signalextracted from the audio signal, the receiver effects activation of arecording apparatus 918 for recording the audio signal transmitted. Inthis manner, a system is provided for radios which makes available amethod comparable to the “VPS” system for television.

According to an additional embodiment of the present invention, a systemis provided making available a data channel operating parallel to theaudio signal, in audio apparatus processing digital data. This datachannel has a low bit rate, and information is introduced into the samein accordance with the method described hereinbefore and extracted fromthe same in accordance with the decoding method described hereinbefore.

It is pointed out that the coder and decoder described herein beforeconstitute just preferred embodiments. The essential features of thecoder for introducing a non-audible data signal into an audio signal aretransforming the audio signal to the spectral range, determining themasking threshold of the audio signal, providing a pseudo-noise signal,providing the data signal, multiplying the pseudo-noise signal by thedata signal so as to provide a frequency-spread data signal, weightingthe spread data signal with the masking threshold, and superimposing theaudio signal and the weighted signal.

The essential features of the decoder for extracting a data signalcontained in an audio signal in non-audible manner, are sampling theaudio signal, non-recursive filtering of the sampled audio signal andcomparing the filtered audio signal to a threshold value so as toretrieve the data signal.

What is claimed is:
 1. A coding method for introducing a data signalinto an audio signal to obtain a combined signal, in which the datasignal is non-audible, said method comprising the following steps: a)converting the audio signal to a spectral representation; b) determininga masking threshold of the audio signal; c) providing a pseudo-noisesignal; d) providing the data signal; e) multiplying the pseudo-noisesignal by the data signal so as to provide a frequency-spread datasignal; f) weighting the frequency-spread data signal with the maskingthreshold to obtain a weighted data signal; and g) superimposing theaudio signal and the weighted data signal to obtain the combined audiosignal.
 2. The coding method of claim 1, wherein step a) includesapplying a fast Fourier transform to the audio signal.
 3. The codingmethod of claim 1, wherein step b) includes the following steps: b1)splitting the spectral representation of the audio signal into criticalbands; b2) determining an energy in each critical band; b3) calculatinga spread function for each critical band; b4) performing a convolutionof spread waveforms of all critical bands with the energies in thecritical bands for obtaining a waveform of an excitation; b5)determining a non-predictability of the audio signal; b6) performing aconvolution of the non-predictability with the spread function to obtaina measure for a tonality; b7) calculating a masking measure on the basisof the tonality; and b8) calculating the masking threshold on the basisof the excitation in consideration of the masking measure.
 4. The codingmethod of claim 1, wherein step b) comprises the following steps: b1)splitting the spectral representation of the audio signal into criticalbands; b2) determining an energy in each critical band; and b3)determining the masking threshold on the basis of energies in thecritical bands in consideration of a masking measure for tonal masking.5. The coding method of claim 1, wherein the pseudo-noise signal has abandwidth of 6 kHz.
 6. The coding method of claim 1, wherein the datasignal has a bandwidth of 50 Hz.
 7. The coding method of claim 1,wherein the data signal is channel-coded by a block code.
 8. The codingmethod of claim 1, wherein, prior to step e), the pseudo-noise signaland the data signal are converted to antipodal signals.
 9. The codingmethod of claim 1, wherein step e) comprises the following steps: e1)performing a BPSK baseband modulation of the data signal with thepseudo-noise signal to obtain a first modulated signal; e2) performing aBPSK modulation of the first modulated signal with a carrier signalhaving a frequency in a range of an audible audio spectrum to obtain asecond modulated signal; and e3) transforming the second modulatedsignal into a spectral domain.
 10. The coding method of claim 9, whereinthe carrier signal is cosinusoidal and has a frequency of 3 kHz.
 11. Thecoding method of claim 9, wherein step e1) includes a step of Manchestercoding of the pseudo-noise signal.
 12. The coding method of claim 1,wherein prior to step g) the weighted data signal of step f) istransformed to a time domain.
 13. The coding method of claim 1, whereinstep g) includes superimposing the audio signal in the spectral domainon the weighted data signal of step f) to obtain a superimposed signaland retransforming the superimposed signal to the time domainthereafter.
 14. The coding method of claim 13, wherein theretransforming to the time domain takes place by a fast Fouriertransform.
 15. A method of decoding a combined audio signal forobtaining a data signal, the combined audio signal including an audiosignal and the data signal, the data signal being multiplied by apseudo-noise signal and weighted with a masking threshold of the audiosignal such that the data signal is contained in a non-audible manner inthe combined audio signal, said method comprising the following steps:a) providing a sampled combined audio signal; b) non-recursive filteringof the sampled combined audio signal using a matched filter, the matchedfilter being matched to the pseudo-noise signal, whereby a filteredcombined audio signal is obtained which includes correlation peaksindicating a correlation between the sampled combined audio signal andthe pseudo-noise signal; and c) comparing the filtered combined audiosignal to a threshold value to detect the peaks, wherein the peaksrepresent the data signal.
 16. The method of claim 9, wherein the audiosignal is received by means of a microphone.
 17. The method of claim 15,wherein prior to step a) the audio signal is lowpass-filtered andamplified.
 18. A coder for introducing a data signal into an audiosignal to obtain a combined signal, in which the data signal isnon-audible, comprising: a converter for converting the audio signal toa spectral representation; a calculator for determining a maskingthreshold of the audio signal; a multiplier for multiplying apseudo-noise signal by the data signal so as to provide afrequency-spread data signal; a weighter for weighting thefrequency-spread data signal with the masking threshold to obtain aweighted data signal; and a superimposer for superimposing the audiosignal and the weighted data signal to obtain the combined audio signal.19. A decoder for decoding a combined audio signal for extracting a datasignal, the combined audio signal including an audio signal and the datasignal, the data signal being multiplied by a pseudo-noise signal andweighted with a masking threshold of the audio signal such that the datasignal is contained in an audio signal in non-audible manner,comprising: a provider for providing a sampled combined audio signal; amatched filter for filtering the sampled audio signal in non-recursivemanner, the matched filter being matched to the pseudo-noise signal,whereby a filtered combined audio signal is obtained which includescorrelation peaks indicating a correlation between the sampled combinedaudio signal and the pseudo-noise signal; and a comparator for comparingthe filtered audio signal to a threshold value to detect the peaks,wherein the peaks represent the data signal.
 20. The decoder of claim19, further comprising a bit synchronization control block for searchingthe filtered combined audio signal for a peak having a certain distancefrom a noise background and for searching for other peaks at distancesfrom the peak, the distances corresponding to a length of thepseudo-noise signal.
 21. The decoder of claim 19, in which the datasignal is organized in frames of several bits, the decoder furthercomprising a frame synchronization block for providing a trigger signalat a beginning of a frame of the data signal.
 22. The decoder of claim21, in which the frame of the data signal is channel encoded, thedecoder further comprising a channel decoder for channel decoding theframe of the data signal to obtain a data word.
 23. A system fordetermining the listener distribution of individual radio stations byway of an identification signal, the identification signal constitutinga data signal, comprising: a coder for introducing the data signal intoan audio signal to obtain a combined signal, in which the data signal isnon-audible, comprising: a converter for converting the audio signal toa spectral representation; a calculator for determining a maskingthreshold of the audio signal; a multiplier for multiplying apseudo-noise signal by the data signal so as to provide afrequency-spread data signal; a weighter for weighting thefrequency-spread data signal with the masking threshold to obtain aweighted data signal; and a superimposer for superimposing the audiosignal and the weighted data signal to obtain the combined audio signal;a decoder for decoding the combined audio signal for extracting the datasignal, the combined audio signal including said audio signal and thedata signal, the data signal being multiplied by said pseudo-noisesignal and weighted with said masking threshold of the audio signal suchthat the data signal is contained in said audio signal in non-audiblemanner, comprising: a provider for providing a sampled combined audiosignal; a matched filter for filtering the sampled audio signal innon-recursive manner, the matched filter being matched to thepseudo-noise signal, whereby a filtered combined audio signal isobtained which includes correlation peaks indicating a correlationbetween the sampled combined audio signal and the pseudo-noise signal;and a comparator for comparing the filtered audio signal to a thresholdvalue to detect the peaks, wherein the peaks represent a receivedidentification signal; and a central station for evaluating the receivedidentification signal.
 24. A system for determining the transmitterreach of a radio station by way of an identification signal, theidentification signal constituting a data signal, comprising: a coderfor introducing the data signal into an audio signal to obtain acombined signal, in which the data signal is non-audible, comprising: aconverter for converting the audio signal to a spectral representation;a calculator for determining a masking threshold of the audio signal; amultiplier for multiplying a pseudo-noise signal by the data signal soas to provide a frequency-spread data signal; a weighter for weightingthe frequency-spread data signal with the masking threshold to obtain aweighted data signal; and a superimposer for superimposing the audiosignal and the weighted data signal to obtain the combined audio signala decoder for decoding the combined audio signal for extracting the datasignal, the combined audio signal including said audio signal and thedata signal, the data signal being multiplied by a pseudo-noise signaland weighted with a masking threshold of the audio signal such that thedata signal is contained in said audio signal in non-audible manner,comprising: a provider for providing a sampled combined audio signal; amatched filter for filtering the sampled audio signal in non-recursivemanner, the matched filter being matched to the pseudo-noise signal,whereby a filtered combined audio signal is obtained which includescorrelation peaks indicating a correlation between the sampled combinedaudio signal and the pseudo-noise signal; and a comparator for comparingthe filtered audio signal to a threshold value to detect the peaks,wherein the peaks represent a received identification signal; and acentral station for evaluating the received identification signal.
 25. Asystem for identifying audio signals with an unequivocal identificationnumber for identifying the sources of copies of sound carriers or soundfiles, the unequivocal identification number constituting a data signal,comprising: a coder for introducing the data signal into an audio signalto obtain a combined signal, in which the data signal is non-audible,comprising: a converter for converting the audio signal to a spectralrepresentation; a calculator for determining a masking threshold of theaudio signal; a multiplier for multiplying a pseudo-noise signal by thedata signal so as to provide a frequency-spread data signal; a weighterfor weighting the frequency-spread data signal with the maskingthreshold to obtain a weighted data signal; and a superimposer forsuperimposing the audio signal and the weighted data signal to obtainthe combined audio signal; a decoder for decoding the combined audiosignal for extracting the data signal, the combined audio signalincluding said audio signal and the data signal, the data signal beingmultiplied by said pseudo-noise signal and weighted with a maskingthreshold of the audio signal such that the data signal is contained insaid audio signal in non-audible manner, comprising: a provider forproviding a sampled combined audio signal; a matched filter forfiltering the sampled audio signal in non-recursive manner, the matchedfilter being matched to the pseudo-noise signal, whereby a filteredcombined audio signal is obtained which includes correlation peaksindicating a correlation between the sampled combined audio signal andthe pseudo-noise signal; and a comparator for comparing the filteredaudio signal to a threshold value to detect the peaks, wherein the peaksrepresent a received unequivocal identification number; and a centralstation for evaluating the received unequivocal identification number.26. A system for the remote control of an audio apparatus by way of acontrol signal, the control signal constituting a data signal,comprising: a coder for introducing the data signal into an audio signalto obtain a combined signal, in which the data signal is non-audible,comprising: a converter for converting the audio signal to a spectralrepresentation; a calculator for determining a masking threshold of theaudio signal; a multiplier for multiplying a pseudo-noise signal by thedata signal so as to provide a frequency-spread data signal; a weighterfor weighting the frequency-spread data signal with the maskingthreshold to obtain a weighted data signal; and a superimposer forsuperimposing the audio signal and the weighted data signal to obtainthe combined audio signal; a decoder for decoding the combined audiosignal for extracting the data signal, the combined audio signalincluding said audio signal and the data signal, the data signal beingmultiplied by said pseudo-noise signal and weighted with said maskingthreshold of the audio signal such that the data signal is contained insaid audio signal in non-audible manner, comprising: a provider forproviding a sampled combined audio signal; a matched filter forfiltering the sampled audio signal in non-recursive manner, the matchedfilter being matched to the pseudo-noise signal, whereby a filteredcombined audio signal is obtained which includes correlation peaksindicating a correlation between the sampled combined audio signal andthe pseudo-noise signal; and a comparator for comparing the filteredaudio signal to a threshold value to detect the peaks, wherein the peaksrepresent the control signal, wherein the audio apparatus is responsiveto the control signal.
 27. The system according to claim 26, in whichrecording of an audio signal in the audio apparatus is started and/orterminated in response to the control signal.
 28. A system for providinga data channel of low bit rate in an audio signal, said channel to beused for transmitting useful data in parallel to the audio signal, theuseful data constituting the data signal, comprising: a coder forintroducing the data signal into an audio signal to obtain a combinedsignal, in which the data signal is non-audible, comprising: a converterfor converting the audio signal to a spectral representation; acalculator for determining a masking threshold of the audio signal; amultiplier for multiplying a pseudo-noise signal by the data signal soas to provide a frequency-spread data signal; a weighter for weightingthe frequency-spread data signal with the masking threshold to obtain aweighted data signal; and a superimposer for superimposing the audiosignal and the weighted data signal to obtain the combined audio signal;a decoder for decoding the combined audio signal for extracting the datasignal, the combined audio signal including said audio signal and thedata signal, the data signal being multiplied by said pseudo-noisesignal and weighted with said masking threshold of the audio signal suchthat the data signal is contained in said audio signal in non-audiblemanner, comprising: a provider for providing a sampled combined audiosignal; a matched filter for filtering the sampled audio signal innon-recursive manner, the matched filter being matched to thepseudo-noise signal, whereby a filtered combined audio signal isobtained which includes correlation peaks indicating a correlationbetween the sampled combined audio signal and the pseudo-noise signal;and a comparator for comparing the filtered audio signal to a thresholdvalue to detect the peaks, wherein the peaks represent the useful data.